This DCHECK is triggered by org.webrtc.PeerConnectionTest#testTrackRemoval.
BUG=webrtc:6465
Review-Url: https://codereview.webrtc.org/2389703002
Cr-Commit-Position: refs/heads/master@{#14481}
This change reduces the number of places where we first fread a I420
frame into a uint8_t buffer, followed by a copy into a frame buffer
object.
BUG=None
Review-Url: https://codereview.webrtc.org/2362683002
Cr-Commit-Position: refs/heads/master@{#14456}
This also makes it possible to drop the RTPFragmentationHeader from
the class VCMEncodedFrame.
BUG=None
Review-Url: https://codereview.webrtc.org/2380933003
Cr-Commit-Position: refs/heads/master@{#14455}
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.
BUG=webrtc:6371
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/2368223002 .
Cr-Commit-Position: refs/heads/master@{#14431}
Also provide a new set of thresholds for the VideoToolbox encoder. The new thresholds were experimentally determined to work well on the iPhone 6S, and also adequately on the iPhone 5S.
BUG=webrtc:5678
Review-Url: https://codereview.webrtc.org/2309743002
Cr-Commit-Position: refs/heads/master@{#14420}
When rtc_build_libyuv=false an incorrect code path
is surfaced in GN.
BUG=webrtc:6412
NOTRY=True
TESTED=gn gen out/foo --args='rtc_build_libyuv=false target_os="ios"'
Review-Url: https://codereview.webrtc.org/2375603002
Cr-Commit-Position: refs/heads/master@{#14392}
This CL removes the use_objc_h264 flag. This means that the VideoToolbox
H264 encoder and decoder will always be built.
BUG=webrtc:4081
NOTRY=TRUE
Review-Url: https://codereview.webrtc.org/2366443003
Cr-Commit-Position: refs/heads/master@{#14372}
"WebRTC.Video.EndToEndDelayInMs"
Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).
BUG=webrtc:6409
Review-Url: https://codereview.webrtc.org/1905563002
Cr-Commit-Position: refs/heads/master@{#14367}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
Deleted from the VideoFrameBuffer base class.
BUG=webrtc:5921
Review-Url: https://codereview.webrtc.org/2278883002
Cr-Commit-Position: refs/heads/master@{#14317}
In the migration to GN templates, some targets got the whole
rtc_common_config removed, which can have unpredicted consequences
in terms of different code behavior due to defines not being set
as expected etc.
It's better to enable this config and only disable the warnings
that fails the build.
BUG=webrtc:6306,webrtc:6307,webrtc:6308
NOTRY=True
Review-Url: https://codereview.webrtc.org/2347263002
Cr-Commit-Position: refs/heads/master@{#14280}
In VP8 the assumption is that a non layer-sync frame depends on all previous
frames on temporal layers less or equal to this frames temporal layer.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2306513003 .
Cr-Commit-Position: refs/heads/master@{#14229}
The MovingAverage class was very specific to the QualityScaler. This
commit generalizes the MovingAverage class to be useful in other
situations as well, and adapts the QualityScaler to use the new
MovingAverage.
BUG=webrtc:6304
Review-Url: https://codereview.webrtc.org/2310853002
Cr-Commit-Position: refs/heads/master@{#14207}
0 means "pause", so forcing it to the min bitrate means we'll never
allow pausing for internal source encoders.
BUG=
Review-Url: https://codereview.webrtc.org/2304603002
Cr-Commit-Position: refs/heads/master@{#14168}
In order to be able to clear out any potentially stashed old frames from
the RtpFrameReferenceFinder we can now clear frames that contain packets
older than |seq_num|.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2304723004
Cr-Commit-Position: refs/heads/master@{#14156}
Changed from 10 to 68.
This is to avoid a flake where the limit is exceeded, see
crbug.com/638554. Our performance tests should flag performance
regressions, we shouldn't rely on crashing because the number of
referenced buffers is not tiny to detect this. However, if a really big
number of buffers (>68) are referenced without being dereferenced it is
likely that we have a bug and frames are leaking in which case we can
DCHECK-crash.
BUG=chromium:638554
Review-Url: https://codereview.webrtc.org/2280593002
Cr-Commit-Position: refs/heads/master@{#14084}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.
Add missing foreman_cif.yuv resource needed for these tests.
For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
If the input to H264VideoToolBoxEncoder is a native CVPixelBuffer and
the quality scaler requests scaling, we fall back to a slow path where
the buffer is converted from NV12 to I420 on the CPU and then uploaded
to a native CVPixelBuffer again. It turns out this scaling is not needed
and that the H264VideoToolBoxEncoder can handle the scaling internally.
BUG=b/30939444
Review-Url: https://codereview.webrtc.org/2258103003
Cr-Commit-Position: refs/heads/master@{#13835}
H.264 frames may have AUD before SPS. This change detects AUD and try to extract SPS and PPS behind AUD.
BUG=webrtc:6173
Review-Url: https://codereview.webrtc.org/2209143002
Cr-Commit-Position: refs/heads/master@{#13765}
Since all FrameObjects have a reference to its PacketBuffer and since
the PacketBuffer can be thrown away at any moment the PacketBuffer
has to be ref counted in order to avoid FrameObjects dereferencing a potentially
destroyed object.
BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2199133004 .
Cr-Commit-Position: refs/heads/master@{#13725}
We need to check the pool to make sure that the encoder session is valid. Otherwise, it appears as if the encoder just does not output frames.
NOTRY=True
BUG=
Review-Url: https://codereview.webrtc.org/2226383002
Cr-Commit-Position: refs/heads/master@{#13698}
Scaling causes us to work the CPU too much, which very quickly degrades quality. This causes us to at least behave better on good networks.
NOTRY=True
BUG=
Review-Url: https://codereview.webrtc.org/2205763002
Cr-Commit-Position: refs/heads/master@{#13630}
